Inbound call is hitting my DID but going to voicemail on Phn

[tanvirvalolok]

Inbound call is hitting my DID, but not reaching my phone, it's going to phone's voicemail. the phone is active and running.

Code: Select all

[Dec 19 19:33:38]     -- Executing [18445011909@trunkinbound:1] AGI("SIP/didforsale_in1-00000002", "agi-DID_route.agi") in new stack
[Dec 19 19:33:38]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Dec 19 19:33:38]     -- <SIP/didforsale_in1-00000002>AGI Script agi-DID_route.agi completed, returning 0
[Dec 19 19:33:38]     -- Executing [068*168*096*145*308@default:1] Goto("SIP/didforsale_in1-00000002", "default,308,1") in new stack
[Dec 19 19:33:38]     -- Goto (default,308,1)
[Dec 19 19:33:38]     -- Executing [308@default:1] Dial("SIP/didforsale_in1-00000002", "SIP/308|60|") in new stack
[Dec 19 19:33:38]   == Using SIP RTP CoS mark 5
[Dec 19 19:33:38] ERROR[12304][C-00000002]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("308|60|", "(null)", ...): Name or service not known
[Dec 19 19:33:38] WARNING[12304][C-00000002]: chan_sip.c:6318 create_addr: No such host: 308|60|
[Dec 19 19:33:38] WARNING[12304][C-00000002]: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Dec 19 19:33:38]   == Everyone is busy/congested at this time (1:0/0/1)
[Dec 19 19:33:38]     -- Executing [308@default:2] Goto("SIP/didforsale_in1-00000002", "default,85026666666666308,1") in new stack
[Dec 19 19:33:38]     -- Goto (default,85026666666666308,1)
[Dec 19 19:33:38]     -- Executing [85026666666666308@default:1] Wait("SIP/didforsale_in1-00000002", "1") in new stack
[Dec 19 19:33:39]     -- Executing [85026666666666308@default:2] VoiceMail("SIP/didforsale_in1-00000002", "308,u") in new stack
[Dec 19 19:33:39]        > 0x7f91a40309e0 -- Strict RTP switching to RTP target address 64.125.111.109:57980 as source
[Dec 19 19:33:39]     -- <SIP/didforsale_in1-00000002> Playing 'vm-theperson.gsm' (language 'en')
[Dec 19 19:33:41]     -- <SIP/didforsale_in1-00000002> Playing 'digits/3.gsm' (language 'en')
[Dec 19 19:33:42]     -- <SIP/didforsale_in1-00000002> Playing 'digits/0.gsm' (language 'en')
[Dec 19 19:33:43]     -- <SIP/didforsale_in1-00000002> Playing 'digits/8.gsm' (language 'en')
[Dec 19 19:33:43]        > 0x7f91a40309e0 -- Strict RTP learning complete - Locking on source address 64.125.111.109:57980
[Dec 19 19:33:43]     -- <SIP/didforsale_in1-00000002> Playing 'vm-isunavail.gsm' (language 'en')
[Dec 19 19:33:45]     -- <SIP/didforsale_in1-00000002> Playing 'vm-intro.gsm' (language 'en')
[Dec 19 19:33:45]   == Spawn extension (default, 85026666666666308, 2) exited non-zero on 'SIP/didforsale_in1-00000002'
[Dec 19 19:33:45] WARNING[12304][C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Dec 19 19:33:45]     -- Executing [h@default:1] AGI("SIP/didforsale_in1-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------)") in new stack
[Dec 19 19:33:45]     -- <SIP/didforsale_in1-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------) com

[ambiorixg12]

[Dec 19 19:33:38] WARNING[12304][C-00000002]: chan_sip.c:6318 create_addr: No such host: 308|60|
[Dec 19 19:33:38] WARNING[12304][C-00000002]: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

SIP device 308 wont be reached as vicidial is working as if you have asterisk 1.4 go to the system settings and change it for the correct version, in order it change | by ,

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3)

the phone is active and running

Code: Select all

sip show peers

is the phone registered and OK?